Release 1.5.21.5.2

5 files changed, 168 insertions, 61 deletions

diff --git a/ChangeLog b/ChangeLog
index 0c991e9..9ec0b11 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,152 @@
+=== release 1.5.2 ===
+
+2015-06-24  Sebastian Dröge <slomo@coaxion.net>
+
+	* configure.ac:
+	  releasing 1.5.2
+
+2015-06-18 13:12:04 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	* gst/rtsp-server/rtsp-client.h:
+	* tests/check/gst/client.c:
+	  rtsp-client: allow application to decide what requirements are supported
+	  Add "check-requirements" signal and vfunc to allow application
+	  (and subclasses) to check the requirements.
+	  Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=749417
+
+2015-06-16 17:50:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* common:
+	  Automatic update of common submodule
+	  From 6015d26 to f74b2df
+
+2015-06-11 17:39:00 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: Always use real payloader when creating streams
+	  A bin that contains the real payloader might be used as payloader. In this
+	  case we have to get the real payloader for the various properties it provides.
+	  Example use cases for this are bins that payload some media and then have
+	  additional elements that add metadata or RTP extension headers to the stream.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750800
+
+2015-06-13 17:14:43 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	  test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
+
+2015-06-12 23:35:32 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	* examples/test-netclock.c:
+	  test-netclock: Use new ntp-time-source property on rtpbin
+	  Select the clock time to be used as NTP time source. This allows proper
+	  synchronization between receivers, independent of sharing base times, and just
+	  requires them to use the same clock.
+
+2015-06-11 20:41:31 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	* examples/test-netclock.c:
+	  test-netclock: Setting the same base time on sender and receiver is not necessary
+	  It's going to be fixed up by rtpbin when using ntp-sync=TRUE
+
+2015-06-11 17:38:52 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 18:10:12 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* docs/libs/gst-rtsp-server.types:
+	  docs: add missing types
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 17:37:25 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* docs/libs/gst-rtsp-server-sections.txt:
+	  docs: add missing apis
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-10 17:14:18 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	  test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
+
+2015-06-05 22:35:39 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+
+	* docs/libs/gst-rtsp-server-sections.txt:
+	* gst/rtsp-server/rtsp-auth.c:
+	* gst/rtsp-server/rtsp-auth.h:
+	  GstRTSPAuth: Add client certificate authentication support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-06-09 13:53:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	  test-netclock-client: Use new GstClock API to wait for clock synchronization
+
+2015-06-09 13:51:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock-client.c:
+	  test-netclock-client: Use a GMainLoop and playbin's source-setup signal
+	  A mainloop is needed to get glimagesink to display something on OSX, and
+	  the source-setup signal just makes things a little bit easier.
+
+2015-06-09 11:30:54 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From d9a3353 to 6015d26
+
+2015-06-08 23:08:34 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From d37af32 to d9a3353
+
+2015-06-07 23:07:31 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 21ba2e5 to d37af32
+
+2015-06-07 17:32:29 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From c408583 to 21ba2e5
+
+2015-06-07 17:06:40 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* docs/libs/Makefile.am:
+	  docs: remove variables that we define in the snippet from common
+	  This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:47 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 44a3517 to c408583
+
+2015-06-07 16:44:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
 === release 1.5.1 ===
 
-2015-06-07  Sebastian Dröge <slomo@coaxion.net>
+2015-06-07 11:20:01 +0200  Sebastian Dröge <sebastian@centricular.com>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
 	* configure.ac:
-	  releasing 1.5.1
+	* gst-rtsp-server.doap:
+	  Release 1.5.1
 
 2015-05-25 16:36:18 +0200  Göran Jönsson <goranjn@axis.com>
 
diff --git a/NEWS b/NEWS
index 6a3a6b7..a6d325d 100644
--- a/NEWS
+++ b/NEWS
@@ -1,2 +1,2 @@
-This is GStreamer RTSP Server 1.5.1
+This is GStreamer RTSP Server 1.5.2
 
diff --git a/RELEASE b/RELEASE
index 5b649a2..fe1d806 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,8 +1,8 @@
 
-Release notes for GStreamer RTSP Server Library 1.5.1
+Release notes for GStreamer RTSP Server Library 1.5.2
 
 
-The GStreamer team is pleased to announce the first release of the unstable
+The GStreamer team is pleased to announce the second release of the unstable
 1.5 release series. The 1.5 release series is adding new features on top of
 the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
 series of the GStreamer multimedia framework. The unstable 1.5 release series
@@ -16,38 +16,11 @@ during the unstable 1.5 release series.
 
   
 
-Features of this release
-    
-
 Bugs fixed in this release
      
-      * 732238 : Listen on the multicast group for RTP/RTCP packets
-      * 734546 : tests: Unref element after usage
-      * 736041 : Protect rtsp transport data.
-      * 736647 : Tunneled RTSP sessions do not always timeout as expected
-      * 737110 : rtsp-client: race condition when closing client connection
-      * 737631 : gst-rtsp-server deadlock while sending response over TCP
-      * 737675 : media: media_unprepare() is kind of broken
-      * 737690 : rtsp-client: deadlock when setting session medias to NULL
-      * 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
-      * 737829 : rtsp-server: deactivate media when shutting down from paused
-      * 738905 : rtsp-client: add stream transport to the context
-      * 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
-      * 740752 : add retransmission support
-      * 740845 : crash when reciving a rtcp after teardown but before client finalize.
-      * 741678 : configure: add --disable-examples switch
-      * 742115 : Examples: Accept a 'port' argument for running multiple instances
-      * 742869 : Remove URI-escaping of RTSP session-id
-      * 742954 : Crash when two treads are in handle_new_sample at the same time.
-      * 743175 : Add support for RECORD
-      * 743346 : When system time is increased the ongoing RTSP sessions will time out.
-      * 743734 : RTCP packets not sent
-      * 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
-      * 745704 : Losing the first packet
-      * 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
-      * 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
-      * 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
-      * 749845 : Client have problem to find the teardown response.
+      * 749417 : rtsp-client: add API to allow application to decide what requirements are supported
+      * 750764 : gst-rtsp-server: add missing apis to doc
+      * 750800 : rtsp-media: always use real payloader when creating streams
 
 ==== Download ====
 
@@ -86,30 +59,11 @@ Applications
   
 Contributors to this release
     
-      * Aleix Conchillo Flaqué
-      * Alistair Buxton
-      * Andreas Frisch
-      * Anila Balavan
-      * Arun Raghavan
-      * Branko Subasic
       * Edward Hervey
-      * Gregor Boirie
-      * Göran Jönsson
       * Hyunjun Ko
-      * Jan Schmidt
-      * Kent-Inge Ingesson
-      * Linus Svensson
-      * Luis de Bethencourt
-      * Matthew Waters
       * Nicolas Dufresne
-      * Nirbheek Chauhan
       * Ognyan Tonchev
-      * Olivier Crête
       * Sebastian Dröge
-      * Sebastian Rasmussen
-      * Srimanta Panda
       * Stefan Sauer
-      * Tim-Philipp Müller
-      * Vincent Penquerc'h
-      * Wim Taymans
+      * Xavier Claessens
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 56a9918..5aa8376 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2,7 +2,7 @@ AC_PREREQ(2.69)
 dnl initialize autoconf
 dnl when going to/from release please set the nano (fourth number) right !
 dnl releases only do Wall, cvs and prerelease does Werror too
-AC_INIT([GStreamer RTSP Server Library], [1.5.1.1],
+AC_INIT([GStreamer RTSP Server Library], [1.5.2],
     [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
     [gst-rtsp-server])
 AG_GST_INIT
@@ -53,13 +53,13 @@ dnl      1.2.5 => 205
 dnl      1.10.9 (who knows) => 1009
 dnl
 dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 501, 0, 501)
+AS_LIBTOOL(GST, 502, 0, 502)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.5.1.1
-GSTPB_REQ=1.5.1.1
-GSTPG_REQ=1.5.1.1
-GSTPD_REQ=1.5.1.1
+GST_REQ=1.5.2
+GSTPB_REQ=1.5.2
+GSTPG_REQ=1.5.2
+GSTPD_REQ=1.5.2
 
 dnl *** autotools stuff ****
 
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index 705191a..e2fd1ab 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.5.2</revision>
+   <branch>1.5</branch>
+   <name></name>
+   <created>2015-06-24</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.2.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.5.1</revision>
    <branch>1.5</branch>
    <name></name>